mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-28 20:53:54 +02:00
8b28aa0767
Both attributes.h and bswap.h have been included from the very beginning of this muxer without there being any reason to do so. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
397 lines
11 KiB
C
397 lines
11 KiB
C
/*
|
|
* Digital Speech Standard (DSS) demuxer
|
|
* Copyright (c) 2014 Oleksij Rempel <linux@rempel-privat.de>
|
|
*
|
|
* This file is part of FFmpeg.
|
|
*
|
|
* FFmpeg is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Lesser General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2.1 of the License, or (at your option) any later version.
|
|
*
|
|
* FFmpeg is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Lesser General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Lesser General Public
|
|
* License along with FFmpeg; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
|
|
#include "libavutil/channel_layout.h"
|
|
#include "libavutil/intreadwrite.h"
|
|
|
|
#include "avformat.h"
|
|
#include "internal.h"
|
|
|
|
#define DSS_HEAD_OFFSET_AUTHOR 0xc
|
|
#define DSS_AUTHOR_SIZE 16
|
|
|
|
#define DSS_HEAD_OFFSET_START_TIME 0x26
|
|
#define DSS_HEAD_OFFSET_END_TIME 0x32
|
|
#define DSS_TIME_SIZE 12
|
|
|
|
#define DSS_HEAD_OFFSET_ACODEC 0x2a4
|
|
#define DSS_ACODEC_DSS_SP 0x0 /* SP mode */
|
|
#define DSS_ACODEC_G723_1 0x2 /* LP mode */
|
|
|
|
#define DSS_HEAD_OFFSET_COMMENT 0x31e
|
|
#define DSS_COMMENT_SIZE 64
|
|
|
|
#define DSS_BLOCK_SIZE 512
|
|
#define DSS_AUDIO_BLOCK_HEADER_SIZE 6
|
|
#define DSS_FRAME_SIZE 42
|
|
|
|
static const uint8_t frame_size[4] = { 24, 20, 4, 1 };
|
|
|
|
typedef struct DSSDemuxContext {
|
|
unsigned int audio_codec;
|
|
int counter;
|
|
int swap;
|
|
int dss_sp_swap_byte;
|
|
int8_t *dss_sp_buf;
|
|
|
|
int packet_size;
|
|
int dss_header_size;
|
|
} DSSDemuxContext;
|
|
|
|
static int dss_probe(const AVProbeData *p)
|
|
{
|
|
if ( AV_RL32(p->buf) != MKTAG(0x2, 'd', 's', 's')
|
|
&& AV_RL32(p->buf) != MKTAG(0x3, 'd', 's', 's'))
|
|
return 0;
|
|
|
|
return AVPROBE_SCORE_MAX;
|
|
}
|
|
|
|
static int dss_read_metadata_date(AVFormatContext *s, unsigned int offset,
|
|
const char *key)
|
|
{
|
|
AVIOContext *pb = s->pb;
|
|
char datetime[64], string[DSS_TIME_SIZE + 1] = { 0 };
|
|
int y, month, d, h, minute, sec;
|
|
int ret;
|
|
|
|
avio_seek(pb, offset, SEEK_SET);
|
|
|
|
ret = avio_read(s->pb, string, DSS_TIME_SIZE);
|
|
if (ret < DSS_TIME_SIZE)
|
|
return ret < 0 ? ret : AVERROR_EOF;
|
|
|
|
if (sscanf(string, "%2d%2d%2d%2d%2d%2d", &y, &month, &d, &h, &minute, &sec) != 6)
|
|
return AVERROR_INVALIDDATA;
|
|
/* We deal with a two-digit year here, so set the default date to 2000
|
|
* and hope it will never be used in the next century. */
|
|
snprintf(datetime, sizeof(datetime), "%.4d-%.2d-%.2dT%.2d:%.2d:%.2d",
|
|
y + 2000, month, d, h, minute, sec);
|
|
return av_dict_set(&s->metadata, key, datetime, 0);
|
|
}
|
|
|
|
static int dss_read_metadata_string(AVFormatContext *s, unsigned int offset,
|
|
unsigned int size, const char *key)
|
|
{
|
|
AVIOContext *pb = s->pb;
|
|
char *value;
|
|
int ret;
|
|
|
|
avio_seek(pb, offset, SEEK_SET);
|
|
|
|
value = av_mallocz(size + 1);
|
|
if (!value)
|
|
return AVERROR(ENOMEM);
|
|
|
|
ret = avio_read(s->pb, value, size);
|
|
if (ret < size) {
|
|
ret = ret < 0 ? ret : AVERROR_EOF;
|
|
goto exit;
|
|
}
|
|
|
|
ret = av_dict_set(&s->metadata, key, value, 0);
|
|
|
|
exit:
|
|
av_free(value);
|
|
return ret;
|
|
}
|
|
|
|
static int dss_read_header(AVFormatContext *s)
|
|
{
|
|
DSSDemuxContext *ctx = s->priv_data;
|
|
AVIOContext *pb = s->pb;
|
|
AVStream *st;
|
|
int ret, version;
|
|
|
|
st = avformat_new_stream(s, NULL);
|
|
if (!st)
|
|
return AVERROR(ENOMEM);
|
|
|
|
version = avio_r8(pb);
|
|
ctx->dss_header_size = version * DSS_BLOCK_SIZE;
|
|
|
|
ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_AUTHOR,
|
|
DSS_AUTHOR_SIZE, "author");
|
|
if (ret)
|
|
return ret;
|
|
|
|
ret = dss_read_metadata_date(s, DSS_HEAD_OFFSET_END_TIME, "date");
|
|
if (ret)
|
|
return ret;
|
|
|
|
ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_COMMENT,
|
|
DSS_COMMENT_SIZE, "comment");
|
|
if (ret)
|
|
return ret;
|
|
|
|
avio_seek(pb, DSS_HEAD_OFFSET_ACODEC, SEEK_SET);
|
|
ctx->audio_codec = avio_r8(pb);
|
|
|
|
if (ctx->audio_codec == DSS_ACODEC_DSS_SP) {
|
|
st->codecpar->codec_id = AV_CODEC_ID_DSS_SP;
|
|
st->codecpar->sample_rate = 11025;
|
|
} else if (ctx->audio_codec == DSS_ACODEC_G723_1) {
|
|
st->codecpar->codec_id = AV_CODEC_ID_G723_1;
|
|
st->codecpar->sample_rate = 8000;
|
|
} else {
|
|
avpriv_request_sample(s, "Support for codec %x in DSS",
|
|
ctx->audio_codec);
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
|
|
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
|
|
st->codecpar->channel_layout = AV_CH_LAYOUT_MONO;
|
|
st->codecpar->channels = 1;
|
|
|
|
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
|
|
st->start_time = 0;
|
|
|
|
/* Jump over header */
|
|
|
|
if (avio_seek(pb, ctx->dss_header_size, SEEK_SET) != ctx->dss_header_size)
|
|
return AVERROR(EIO);
|
|
|
|
ctx->counter = 0;
|
|
ctx->swap = 0;
|
|
|
|
ctx->dss_sp_buf = av_malloc(DSS_FRAME_SIZE + 1);
|
|
if (!ctx->dss_sp_buf)
|
|
return AVERROR(ENOMEM);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void dss_skip_audio_header(AVFormatContext *s, AVPacket *pkt)
|
|
{
|
|
DSSDemuxContext *ctx = s->priv_data;
|
|
AVIOContext *pb = s->pb;
|
|
|
|
avio_skip(pb, DSS_AUDIO_BLOCK_HEADER_SIZE);
|
|
ctx->counter += DSS_BLOCK_SIZE - DSS_AUDIO_BLOCK_HEADER_SIZE;
|
|
}
|
|
|
|
static void dss_sp_byte_swap(DSSDemuxContext *ctx,
|
|
uint8_t *dst, const uint8_t *src)
|
|
{
|
|
int i;
|
|
|
|
if (ctx->swap) {
|
|
for (i = 3; i < DSS_FRAME_SIZE; i += 2)
|
|
dst[i] = src[i];
|
|
|
|
for (i = 0; i < DSS_FRAME_SIZE - 2; i += 2)
|
|
dst[i] = src[i + 4];
|
|
|
|
dst[1] = ctx->dss_sp_swap_byte;
|
|
} else {
|
|
memcpy(dst, src, DSS_FRAME_SIZE);
|
|
ctx->dss_sp_swap_byte = src[DSS_FRAME_SIZE - 2];
|
|
}
|
|
|
|
/* make sure byte 40 is always 0 */
|
|
dst[DSS_FRAME_SIZE - 2] = 0;
|
|
ctx->swap ^= 1;
|
|
}
|
|
|
|
static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt)
|
|
{
|
|
DSSDemuxContext *ctx = s->priv_data;
|
|
AVStream *st = s->streams[0];
|
|
int read_size, ret, offset = 0, buff_offset = 0;
|
|
int64_t pos = avio_tell(s->pb);
|
|
|
|
if (ctx->counter == 0)
|
|
dss_skip_audio_header(s, pkt);
|
|
|
|
if (ctx->swap) {
|
|
read_size = DSS_FRAME_SIZE - 2;
|
|
buff_offset = 3;
|
|
} else
|
|
read_size = DSS_FRAME_SIZE;
|
|
|
|
ctx->counter -= read_size;
|
|
ctx->packet_size = DSS_FRAME_SIZE - 1;
|
|
|
|
ret = av_new_packet(pkt, DSS_FRAME_SIZE);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
pkt->duration = 264;
|
|
pkt->pos = pos;
|
|
pkt->stream_index = 0;
|
|
s->bit_rate = 8LL * ctx->packet_size * st->codecpar->sample_rate * 512 / (506 * pkt->duration);
|
|
|
|
if (ctx->counter < 0) {
|
|
int size2 = ctx->counter + read_size;
|
|
|
|
ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset,
|
|
size2 - offset);
|
|
if (ret < size2 - offset)
|
|
goto error_eof;
|
|
|
|
dss_skip_audio_header(s, pkt);
|
|
offset = size2;
|
|
}
|
|
|
|
ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset,
|
|
read_size - offset);
|
|
if (ret < read_size - offset)
|
|
goto error_eof;
|
|
|
|
dss_sp_byte_swap(ctx, pkt->data, ctx->dss_sp_buf);
|
|
|
|
if (ctx->dss_sp_swap_byte < 0) {
|
|
ret = AVERROR(EAGAIN);
|
|
goto error_eof;
|
|
}
|
|
|
|
return pkt->size;
|
|
|
|
error_eof:
|
|
av_packet_unref(pkt);
|
|
return ret < 0 ? ret : AVERROR_EOF;
|
|
}
|
|
|
|
static int dss_723_1_read_packet(AVFormatContext *s, AVPacket *pkt)
|
|
{
|
|
DSSDemuxContext *ctx = s->priv_data;
|
|
AVStream *st = s->streams[0];
|
|
int size, byte, ret, offset;
|
|
int64_t pos = avio_tell(s->pb);
|
|
|
|
if (ctx->counter == 0)
|
|
dss_skip_audio_header(s, pkt);
|
|
|
|
/* We make one byte-step here. Don't forget to add offset. */
|
|
byte = avio_r8(s->pb);
|
|
if (byte == 0xff)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
size = frame_size[byte & 3];
|
|
|
|
ctx->packet_size = size;
|
|
ctx->counter -= size;
|
|
|
|
ret = av_new_packet(pkt, size);
|
|
if (ret < 0)
|
|
return ret;
|
|
pkt->pos = pos;
|
|
|
|
pkt->data[0] = byte;
|
|
offset = 1;
|
|
pkt->duration = 240;
|
|
s->bit_rate = 8LL * size * st->codecpar->sample_rate * 512 / (506 * pkt->duration);
|
|
|
|
pkt->stream_index = 0;
|
|
|
|
if (ctx->counter < 0) {
|
|
int size2 = ctx->counter + size;
|
|
|
|
ret = avio_read(s->pb, pkt->data + offset,
|
|
size2 - offset);
|
|
if (ret < size2 - offset) {
|
|
av_packet_unref(pkt);
|
|
return ret < 0 ? ret : AVERROR_EOF;
|
|
}
|
|
|
|
dss_skip_audio_header(s, pkt);
|
|
offset = size2;
|
|
}
|
|
|
|
ret = avio_read(s->pb, pkt->data + offset, size - offset);
|
|
if (ret < size - offset) {
|
|
av_packet_unref(pkt);
|
|
return ret < 0 ? ret : AVERROR_EOF;
|
|
}
|
|
|
|
return pkt->size;
|
|
}
|
|
|
|
static int dss_read_packet(AVFormatContext *s, AVPacket *pkt)
|
|
{
|
|
DSSDemuxContext *ctx = s->priv_data;
|
|
|
|
if (ctx->audio_codec == DSS_ACODEC_DSS_SP)
|
|
return dss_sp_read_packet(s, pkt);
|
|
else
|
|
return dss_723_1_read_packet(s, pkt);
|
|
}
|
|
|
|
static int dss_read_close(AVFormatContext *s)
|
|
{
|
|
DSSDemuxContext *ctx = s->priv_data;
|
|
|
|
av_freep(&ctx->dss_sp_buf);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int dss_read_seek(AVFormatContext *s, int stream_index,
|
|
int64_t timestamp, int flags)
|
|
{
|
|
DSSDemuxContext *ctx = s->priv_data;
|
|
int64_t ret, seekto;
|
|
uint8_t header[DSS_AUDIO_BLOCK_HEADER_SIZE];
|
|
int offset;
|
|
|
|
if (ctx->audio_codec == DSS_ACODEC_DSS_SP)
|
|
seekto = timestamp / 264 * 41 / 506 * 512;
|
|
else
|
|
seekto = timestamp / 240 * ctx->packet_size / 506 * 512;
|
|
|
|
if (seekto < 0)
|
|
seekto = 0;
|
|
|
|
seekto += ctx->dss_header_size;
|
|
|
|
ret = avio_seek(s->pb, seekto, SEEK_SET);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
avio_read(s->pb, header, DSS_AUDIO_BLOCK_HEADER_SIZE);
|
|
ctx->swap = !!(header[0] & 0x80);
|
|
offset = 2*header[1] + 2*ctx->swap;
|
|
if (offset < DSS_AUDIO_BLOCK_HEADER_SIZE)
|
|
return AVERROR_INVALIDDATA;
|
|
if (offset == DSS_AUDIO_BLOCK_HEADER_SIZE) {
|
|
ctx->counter = 0;
|
|
offset = avio_skip(s->pb, -DSS_AUDIO_BLOCK_HEADER_SIZE);
|
|
} else {
|
|
ctx->counter = DSS_BLOCK_SIZE - offset;
|
|
offset = avio_skip(s->pb, offset - DSS_AUDIO_BLOCK_HEADER_SIZE);
|
|
}
|
|
ctx->dss_sp_swap_byte = -1;
|
|
return 0;
|
|
}
|
|
|
|
|
|
AVInputFormat ff_dss_demuxer = {
|
|
.name = "dss",
|
|
.long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard (DSS)"),
|
|
.priv_data_size = sizeof(DSSDemuxContext),
|
|
.read_probe = dss_probe,
|
|
.read_header = dss_read_header,
|
|
.read_packet = dss_read_packet,
|
|
.read_close = dss_read_close,
|
|
.read_seek = dss_read_seek,
|
|
.extensions = "dss"
|
|
};
|